• Electronics and Telecommunications Trends
• /
• v.17 no.2 s.74
• /
• pp.7-17
• /
• 2002

오늘날 대부분의 신기술은 통신규제를 우회하므로, 각국 정부는 규제를 포기하고 기술과 시장이 통신업종의 발전을 지향하는 방향으로 나아가고 있다. 그러나, IP Telephony와 관련된 최근의 동향은 기존의 법률과 규제가 각국에서 IP Telephony의 발전을 결정하는 강력한 요소임을 보여주고 있다. 본 고는 IP Telephony와 관련한 기술발전 추세와 표준화동향, 시장도입효과를 알아보고 특히, IP Telephony와 관련한 구체적인 도입방안을 위한 논의가 있었던 유럽연합과 프랑스를 중심으로 규제동향을 살펴보았다.

• Journal of the Korea Safety Management & Science
• /
• v.11 no.3
• /
• pp.199-207
• /
• 2009

IP Telephony service was restricted to an outgoing call and low quality since the trust domestic IP Telephony service launch of Saerome co. Ltd, in Jan. 2000. However, Interest of IP Telephony service, which is substituted for PSTN, has been highly elated because of the developed equipment softswitch and new technology. This kind of importance and marketing of VoIP are recognized to telecommunication providers. With this trend, they try to administrate customer satisfaction and invest R&D to survive in this hard competition and unexpected change. To achieve this objective, they should try to realize the searching process of the quality decision attribution (QDA). However, there is little research on the aspect of service quality of Internet telephony so far. For this, the investigator established the tangibles, the reliability, the responsibility, the assurance, the empathy, the charge with information sources as core elements. In order to examine the influence of IP Telephony service upon the attitudes toward a brand and the purchase intention.

• Journal of Advanced Marine Engineering and Technology
• /
• v.27 no.1
• /
• pp.65-75
• /
• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• Journal of the Korea Society of Computer and Information
• /
• v.10 no.5 s.37
• /
• pp.237-244
• /
• 2005

IP-Internet Telephony Service has not vet been achieved that of operating an IP-PBX service and a consumer Internet telephone services using VoIP technologies. In this paper, i suggest that the technologies of the VoIP Secure Gateway have connecting and securing for IP-Internet Telephony Service which makes If telephony protocols, firewall VPN tunneling, using Application Level Gateway, connection of the VoIP Secure Gateway. I suggest of telecommunication technologies that are enables an enterprise If-PBX service to interoperate with a consumer IP telephony service through a firewall. Also, I have proposed the solutions of security problems which was the security for VoIP Secure Gateway.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.11 no.4
• /
• pp.83-89
• /
• 2011

Nowadays, Internet Telephony Services based on SIP are gaining an explosive increase for general user. Thus a various demands of the user about convenience of IP-PBX are growing. The most important requirements of these is hybrid IP telephony system which is connected not only internet telephone but also general PSTN telephone. Therefore, in this paper, we have developed Hybrid IP-PBX connected all type of telephony terminals for small office. We have developed FXS module for connecting all type and FXO for using PSTN telephone number also. This Hybrid IP-PBX that is can be connected Soft-phone provide various optional services. We measured voice quality and test simultaneous calling for proof of validity.

• Journal of Information Technology Applications and Management
• /
• v.17 no.2
• /
• pp.133-154
• /
• 2010

Recently, Internet Telephony has become increasingly popular in telecommunication industry. However, previous research on Internet Telephony has focused on analyzing specific Internet Telephonysolutions, identifyingthe Internet Telephony movement itself. The research on prediction models about Internet Telephony adoption has been minimal. The main propose of this study is to develop models for predicting transition intention from using traditional telephones to using Internet Telephony. To do so, this study uses data mining methods to analyze demands in the IT communications market and to provide management strategies for Internet telephony providers. Especially this study uses discriminant analysis, logistic regression, classification tree, and neural nets to develop those prediction models toward Internet Telephony adoption. The models are compared with each other and a superior model is chosen.

• Journal of Digital Convergence
• /
• v.10 no.6
• /
• pp.259-267
• /
• 2012

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented an Internet telephony network system which is developed by using Asterisk and open source softwares. It is developed on the linux system and has some features such as VoIP telephony service between SIP phones, voice mail, and call recording. It also supports web-based functions such as SIP users and server system management that is implemented by Apache web server and PHP programs. Afterwards, this system will be applied as VoIP network base technology for small sized companies and organizations. It will paly a role for encouraging companies to use open source softwares.

• Journal of KIISE:Computing Practices and Letters
• /
• v.9 no.1
• /
• pp.47-59
• /
• 2003

The Internet telephony service is one of the commercially successful Internet application services. VoIP technology makes the service come true. VoIP deploys H.323 or SIP as the standard protocol for the distributed multimedia services over the Internet in which QoS is not guaranteed. VoIP carries the packetized voice over the RTP/UDP/IP protocol stack. The data transmission trouble is caused by UDP when the service is provided in private networks and some ISP-provided Internet access networks in the private address space. The Internet telephony users in such networks cannot listen the voices of the other parties in the public Internet or PSTN. Making the problem more difficult, the Internet telephony service considered in this paper gets the incoming voice packets of every session through only one UDP port number. In this paper, three schemes including the terminal proxy, the gateway proxy, and the protocol translation are suggested to solve the problems. The design and implementation of the NAT proxy server based on gateway proxy scheme are described in detail.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.9C
• /
• pp.842-852
• /
• 2002

The fast advance in the VoIP technologies gives a rich opportunity to create different kind of VoIP applications such as IP telephony services. The application level call signaling protocols such as ITU-T H.323 and IETF SIP provide the communication functions of end-to-end call setup and release. Currently, there is a lot of H.323 based VoIP products in the market, however SIP is considered as a suitable protocol for supporting applications in IP environments, so SIP based VoIP products and services begin to appear. In this paper, firstly we present the characteristics of some possible SIP based applications and describe the design and implementation of a VoIP example service named PC-to-PC Internet telephony service using the developed SIP network components. The PC-to-PC Internet telephony service and User Agent are developed in MS window 98/2000 using visual C/C++, and Proxy server and Registrar in Linux 7.0 using C, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.11 no.2
• /
• pp.293-300
• /
• 2007

As the development of the various IP-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN-based telephony service. But there are many issues resolved to spread the Internet telephony service. One of them is to support the emergency services in the Internet telephony. In the case of USA, it has been regulated that 911 services should be supported in the Internet telephony services using VoIP on the similar performance level to PSTN 911 service. According to the regulation, basic VoIP 911 calls should be routed to the general access line of LEA without the location information or the callback number, but the enhanced VoIP 911 calls with the location information and callback number should be routed on the dedicated 911 network and destined to the local 911 distribution center such as PSAP. But, in the current VoIP-based Internet telephony network, the emergency call service has not been handled as one of the special services as well at has a worse performance in comparison to it on PSTN. Moreover, the service has a critical problem that it can not be destined to the nearest PSAP because of the insufficient information about the location information and the call back number. In this paper, we suggest the SIP-based emergency call service mechanism in order to resolve the problems above mentioned. This suggested mechanism is implemented to show its effectiveness and efficiency.